[asterisk-users] Asterisk does not accepts SIP registration
Yaroslav Panych
panych.y at gmail.com
Tue Oct 25 06:30:53 CDT 2011
Hello
Always returns 401 Unauthorized, because of
[Oct 25 11:59:48] NOTICE[2501] chan_sip.c: Correct auth, but based on
stale nonce received from '"L6"
<sip:L6 at 172.30.8.13:5060>;tag=31b9dc9e-684902'
L6 is realtime device of type FRIEND (DLINK DVG7022S)
Reviewed SIP conversation - no results.
SIP debug
<--- SIP read from UDP:172.30.8.18:5060 --->
REGISTER sip:172.30.8.13:5060 SIP/2.0
v:SIP/2.0/UDP 172.30.8.18:5060;branch=z9hG4bKe3a0c285855ae0e5
f:"L6" <sip:L6 at 172.30.8.13:5060>;tag=31b9dc9e-684902
t:"L6" <sip:L6 at 172.30.8.13:5060>
i:BD2F-1923-466848179B9BEAA6258E-001 at SipHost
CSeq:23 REGISTER
m:<sip:L6 at 172.30.8.18:5060>
Expires:0
Max-Forwards:70
User-Agent:dlink 12-36-9924913
l:0
<------------->
<--- Transmitting (no NAT) to 172.30.8.18:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
172.30.8.18:5060;branch=z9hG4bKe3a0c285855ae0e5;received=172.30.8.18
From: "L6" <sip:L6 at 172.30.8.13:5060>;tag=31b9dc9e-684902
To: "L6" <sip:L6 at 172.30.8.13:5060>;tag=as1a9dabcb
Call-ID: BD2F-1923-466848179B9BEAA6258E-001 at SipHost
CSeq: 23 REGISTER
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1555e540"
Content-Length: 0
<------------>
REGISTER sip:172.30.8.13:5060 SIP/2.0
v:SIP/2.0/UDP 172.30.8.18:5060;branch=z9hG4bK629009e8cee7f7e3
f:"L6" <sip:L6 at 172.30.8.13:5060>;tag=31b9dc9e-684902
t:"L6" <sip:L6 at 172.30.8.13:5060>
i:BD2F-1923-466848179B9BEAA6258E-001 at SipHost
CSeq:24 REGISTER
m:<sip:L6 at 172.30.8.18:5060>
Expires:0
Max-Forwards:70
Authorization:Digest
username="L6",realm="asterisk",nonce="1555e540",uri="sip:172.30.8.13:5060",response="f47b23120619eb9e6d184bafa48b92c9",algorithm=MD5
User-Agent:dlink 12-36-9924913
l:0
<------------->
[Oct 25 11:59:48] NOTICE[2501] chan_sip.c: Correct auth, but based on
stale nonce received from '"L6"
<sip:L6 at 172.30.8.13:5060>;tag=31b9dc9e-684902'
[Oct 25 11:59:48] VERBOSE[2501] chan_sip.c:
<--- Transmitting (no NAT) to 172.30.8.18:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
172.30.8.18:5060;branch=z9hG4bK629009e8cee7f7e3;received=172.30.8.18
From: "L6" <sip:L6 at 172.30.8.13:5060>;tag=31b9dc9e-684902
To: "L6" <sip:L6 at 172.30.8.13:5060>;tag=as014cd348
Call-ID: BD2F-1923-466848179B9BEAA6258E-001 at SipHost
CSeq: 24 REGISTER
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="11195a41", stale=true
Content-Length: 0
<------------>
sip.conf
[general]
context = default
allowguest = no
bindport = 5060
bindaddr = 0.0.0.0
allowexternaldomains = no
allowoverlap = yes
allowsubscribe = yes
allowtransfer = yes
alwaysauthreject = no
autodomain = no
callevents = no
canreinvite = no
checkmwi = 10
compactheaders = no
defaultexpiry = 120
domain=sop-korniychuk
domain=172.30.8.13
domain=172.30.8.13:5060
dumphistory = no
externrefresh = 10
g726nonstandard = no
notifyringing = yes
srvlookup = yes
t1min = 100
t38pt_udptl = no
;tos_audio = none
;tos_sip = none
;tos_video = none
trustrpid = no
useragent = Asterisk PBX
usereqphone = no
videosupport = no
disallow = all
allow = alaw
type = friend
host=dynamic
context = noop-context
dtmfmode=rfc2833
;language = ru
;sipdebug=yes
nat=no
rtcachefriends=yes
qualify=10000
deny=0.0.0.0/0.0.0.0
permit=172.30.8.0/255.255.255.0
sip show settings
Global Settings:
----------------
UDP Bindaddress: 0.0.0.0:5060
TCP SIP Bindaddress: Disabled
TLS SIP Bindaddress: Disabled
Videosupport: No
Textsupport: No
Ignore SDP sess. ver.: No
AutoCreate Peer: No
Match Auth Username: No
Allow unknown access: No
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promisc. redir: No
Enable call counters: No
SIP domain support: Yes
Realm. auth: No
Our auth realm asterisk
Use domains as realms: No
Call to non-local dom.: No
URI user is phone no: No
Always auth rejects: No
Direct RTP setup: No
User Agent: Asterisk PBX
SDP Session Name: Asterisk PBX 1.8.5.0
SDP Owner Name: root
Reg. context: (not set)
Regexten on Qualify: No
Legacy userfield parse: No
Caller ID: asterisk
From: Domain:
Record SIP history: Off
Call Events: Off
Auth. Failure Events: Off
T.38 support: No
T.38 EC mode: Unknown
T.38 MaxDtgrm: -1
SIP realtime: Enabled
Qualify Freq : 60000 ms
Q.850 Reason header: No
Network QoS Settings:
---------------------------
IP ToS SIP: CS0
IP ToS RTP audio: CS0
IP ToS RTP video: CS0
IP ToS RTP text: CS0
802.1p CoS SIP: 4
802.1p CoS RTP audio: 5
802.1p CoS RTP video: 6
802.1p CoS RTP text: 5
Jitterbuffer enabled: No
Network Settings:
---------------------------
SIP address remapping: Disabled, no localnet list
Externhost: <none>
externaddr: (null)
Externrefresh: 10
Global Signalling Settings:
---------------------------
Codecs: 0x8 (alaw)
Codec Order: alaw:20
Relax DTMF: No
RFC2833 Compensation: No
Symmetric RTP: No
Compact SIP headers: No
RTP Keepalive: 0 (Disabled)
RTP Timeout: 0 (Disabled)
RTP Hold Timeout: 0 (Disabled)
MWI NOTIFY mime type: application/simple-message-summary
DNS SRV lookup: Yes
Pedantic SIP support: Yes
Reg. min duration 60 secs
Reg. max duration: 3600 secs
Reg. default duration: 120 secs
Outbound reg. timeout: 20 secs
Outbound reg. attempts: 0
Notify ringing state: Yes
Include CID: No
Notify hold state: No
SIP Transfer mode: open
Max Call Bitrate: 384 kbps
Auto-Framing: No
Outb. proxy: <not set>
Session Timers: Accept
Session Refresher: uas
Session Expires: 1800 secs
Session Min-SE: 90 secs
Timer T1: 500
Timer T1 minimum: 100
Timer B: 32000
No premature media: Yes
Max forwards: 70
Default Settings:
-----------------
Allowed transports: UDP
Outbound transport: UDP
Context: noop-context
Force rport: No
DTMF: rfc2833
Qualify: 10000
Use ClientCode: No
Progress inband: Never
Language:
MOH Interpret: default
MOH Suggest:
Voice Mail Extension: asterisk
Realtime SIP Settings:
----------------------
Realtime Peers: Yes
Realtime Regs: No
Cache Friends: Yes
Update: Yes
Ignore Reg. Expire: No
Save sys. name: No
Auto Clear: 120 (Disabled)
----
When registering soft SIP client - all okay.
What I'm doing wrong?
regards, Yaroslav.
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